Legal claims defining the scope of protection. Each claim is shown in both the original legal language and a plain English translation.
1. A media system comprising a first playback device and a second playback device, the media system comprising: the first playback device comprising: one or more microphones; a network interface; a processor; a non-transitory computer-readable medium; program instructions stored on the non-transitory computer-readable medium that, when executed by the processor, cause the first playback device to perform functions comprising: receiving an indication of an available amount of computational power of the second playback device; and based on the received indication of the available amount of computational power of the second playback device, sending, to the second playback device, a set of audio signals received by the one or more microphones of the first playback device, wherein the set of audio signals includes at least one voice input; the second playback device comprising: a network interface; a processor; a non-transitory computer-readable medium; program instructions stored on the non-transitory computer-readable medium that, when executed by the processor, cause the second playback device to perform functions comprising: receiving, from the first playback device, the set of audio signals received by the one or more microphones of the first playback device; processing the set of audio signals using a first set of audio processing algorithms to determine a set of signal measures corresponding to the set of audio signals; based on the set of signal measures, identifying, from the set of audio signals, at least two audio signals that are to be re-processed using a second set of audio processing algorithms so as to improve the respective signal measures of the at least two audio signals; re-processing the at least two audio signals using the second set of audio processing algorithms; combining the re-processed at least two audio signals into a combined audio signal; and sending the combined audio signal to a network device.
A media system includes a first playback device with microphones and a second playback device connected via a network. The first device captures audio signals, including voice inputs, and sends them to the second device based on its available computational power. The second device processes the audio signals using initial algorithms to measure signal quality, identifies signals needing improvement, and re-processes them with advanced algorithms. The improved signals are combined and sent to a network device. This system distributes audio processing tasks between devices to enhance signal quality efficiently, leveraging computational resources where available. The first device offloads processing to the second device when it has sufficient capacity, optimizing performance without requiring all processing to occur locally. The second device dynamically selects and re-processes specific audio segments to improve clarity or other signal metrics before combining them for output. This approach enables real-time audio enhancement in distributed environments, such as smart home systems or multi-device setups.
2. The media system of claim 1 , wherein the network device comprises a server, the server comprising: a network interface; a processor; a non-transitory computer-readable medium; program instructions stored on the non-transitory computer-readable medium that are executable by the processor to cause the server to perform functions comprising: determining the available amount of computational power of the second playback device; and sending, to the first playback device, the indication of the available amount of computational power of the second playback device.
A media system includes a network device, such as a server, that facilitates media playback across multiple playback devices. The server determines the available computational power of a second playback device and communicates this information to a first playback device. This allows the system to dynamically allocate media processing tasks based on the computational capabilities of the devices involved. The server includes a network interface for communication, a processor for executing instructions, and a non-transitory computer-readable medium storing program instructions. These instructions enable the server to assess the computational resources of the second playback device and transmit this data to the first playback device. The system optimizes media playback by leveraging the available processing power of connected devices, ensuring efficient distribution of workloads. This approach enhances performance and scalability in distributed media playback environments.
3. The media system of claim 1 , wherein the second playback device further comprises program instructions stored on the non-transitory computer-readable medium that are executable by the processor to cause the second playback device to perform functions comprising: determining the available amount of computational power of the second playback device; and sending, to the first playback device, the indication of the available amount of computational power of the second playback device.
This invention relates to a media system with distributed computational capabilities among playback devices. The system addresses the problem of efficiently utilizing computational resources across multiple playback devices to enhance media processing, such as rendering, encoding, or decoding tasks. The system includes at least two playback devices, each with a processor and a non-transitory computer-readable medium storing program instructions. The first playback device is configured to send a request for computational assistance to the second playback device. The second playback device includes additional program instructions that enable it to determine its available computational power, such as processing capacity, memory availability, or other performance metrics. The second playback device then sends an indication of this available computational power back to the first playback device. This allows the first playback device to dynamically allocate tasks based on the computational resources available in the system, improving efficiency and performance. The system may be used in applications like multi-device media rendering, distributed video encoding, or real-time media processing where computational load balancing is beneficial.
4. The media system of claim 1 , the second playback device further comprising one or more microphones.
A media system enables synchronized playback of audio content across multiple playback devices, such as speakers, to create an immersive listening experience. The system addresses the challenge of maintaining audio synchronization and spatial accuracy in multi-device environments, particularly in scenarios where devices may be dynamically added or removed. The system includes a primary playback device that coordinates playback timing and a secondary playback device that receives and processes audio signals for synchronized output. The secondary playback device further includes one or more microphones to capture ambient sound or user input, enabling interactive features such as voice commands, environmental adaptation, or feedback-based adjustments. The microphones may be used to detect acoustic conditions, adjust playback parameters, or facilitate communication between devices. The system ensures that audio content remains synchronized while allowing dynamic interaction through the secondary device's microphones, enhancing user engagement and system adaptability.
5. The media system of claim 4 , wherein at least a portion of the set of audio signals received by the one or more microphones of the first playback device is additionally received by the one or more microphones of the second playback device.
This invention relates to a media system for synchronized audio playback across multiple playback devices, addressing the challenge of maintaining audio coherence when devices are positioned at different distances from a sound source. The system includes at least two playback devices, each equipped with one or more microphones. The microphones capture audio signals from a sound source, such as a user's voice or an external audio source. The system processes these signals to determine timing offsets between the devices, compensating for differences in signal arrival times due to varying distances from the source. This compensation ensures synchronized playback of audio content across the devices, enhancing audio quality and user experience. The system may also use the captured audio signals for additional functions, such as voice recognition or environmental noise analysis. By leveraging multiple microphones across devices, the system improves robustness and accuracy in audio processing, particularly in scenarios where devices are not uniformly spaced or oriented relative to the sound source. The invention aims to provide seamless, synchronized audio playback in multi-device environments, overcoming limitations of traditional single-device systems.
6. The media system of claim 1 , the media system further comprising a third playback device, and wherein the first playback device further comprises program instructions stored on the non-transitory computer-readable medium that are executable by the processor to cause the first playback device to perform functions comprising: receiving an indication of an available amount of computational power of the third playback device; comparing the available amount of computational power of the second playback device with the available amount of computational power of the third playback device; and based on comparing the available amount of computational power of the second playback device with the available amount of computational power of the third playback device, deciding to send the set of audio signals received by the one or more microphones of the first playback device to the second playback device.
A media system includes multiple playback devices that collaborate to process and play audio signals. The system addresses the challenge of efficiently distributing audio processing tasks among devices with varying computational capabilities. One playback device, equipped with microphones, captures audio signals and evaluates the available computational power of other playback devices in the system. By comparing the computational resources of these devices, the system dynamically decides which device should handle the audio processing tasks. This ensures optimal performance by offloading processing to the most capable device, improving overall system efficiency and responsiveness. The system may include additional playback devices, and the decision-making process considers their computational availability to determine the best device for processing the captured audio signals. This approach enhances the scalability and adaptability of the media system in environments with varying device capabilities.
7. The media system of claim 1 , wherein each of the first set of audio processing algorithms and the second set of audio processing algorithms comprises at least one of an echo cancellation algorithm or a beamforming algorithm.
This invention relates to a media system designed to enhance audio processing in communication devices. The system addresses the problem of poor audio quality in real-time communication, such as video calls or voice calls, by dynamically selecting and applying different sets of audio processing algorithms based on environmental conditions. The system includes a first set of audio processing algorithms optimized for one set of conditions and a second set optimized for another. The system monitors environmental factors, such as background noise or speaker positions, and switches between the two algorithm sets to improve audio clarity. The invention ensures that the selected algorithms are tailored to the current environment, reducing echo and improving directional audio capture. The system may include algorithms like echo cancellation, which removes unwanted reflections, and beamforming, which focuses on a specific sound source while suppressing background noise. By dynamically adjusting the audio processing, the system provides clearer and more reliable audio in various communication scenarios.
8. The media system of claim 1 , wherein the first playback device further comprises program instructions stored on the non-transitory computer-readable medium that are executable by the processor to cause the first playback device to perform functions comprising: based on the received indication of the available amount of computational power of the second playback device, determining that the available amount of computational power of the second playback device is above a threshold amount of computational power.
A media system includes multiple playback devices that collaborate to process and render media content. The system addresses the challenge of efficiently distributing computational tasks between devices with varying processing capabilities to optimize performance and resource utilization. One playback device receives an indication of the available computational power of a second playback device. The system evaluates whether the second device's available computational power exceeds a predefined threshold. If the threshold is met, the system may offload or distribute certain processing tasks to the second device, leveraging its excess capacity to enhance overall system performance. This dynamic allocation ensures that media playback remains smooth and responsive, even when individual devices have limited processing resources. The system may also include features for synchronizing playback across devices and managing network latency to maintain high-quality audio or video output. The invention improves media playback efficiency by intelligently distributing computational workloads based on real-time device capabilities.
9. The media system of claim 1 , wherein the at least one voice input comprises at least one command, and wherein the network device comprises a server, the server comprising: a network interface; a processor; a non-transitory computer-readable medium; program instructions stored on the non-transitory computer-readable medium that are executable by the processor to cause the server to perform functions comprising: receiving the combined audio signal from the second playback device; based on the combined audio signal, recognizing the at least one command; and transmitting an indication of the recognized at least one command to the first playback device.
This invention relates to a media system for processing voice commands in a multi-device audio environment. The system addresses the challenge of accurately capturing and interpreting voice inputs when multiple playback devices are active, ensuring seamless command recognition across distributed audio systems. The media system includes at least two playback devices and a network device, such as a server, that processes voice commands. One playback device captures a voice input, such as a command, and combines it with an audio signal from another playback device. The combined signal is sent to the server, which receives and analyzes the signal to recognize the command. The server then transmits an indication of the recognized command to the first playback device, enabling appropriate action, such as adjusting playback settings or executing a specific function. The server includes a network interface for communication, a processor for executing instructions, and a non-transitory computer-readable medium storing program instructions. These instructions enable the server to receive the combined audio signal, process it to identify the command, and relay the recognized command back to the originating playback device. This ensures that voice commands are accurately interpreted even when multiple audio sources are active, improving usability in multi-device audio setups.
10. The media system of claim 1 , wherein the at least two audio signals comprise audio signals having signal to noise ratios meeting a threshold amount.
The media system processes multiple audio signals to enhance audio quality. The system captures at least two audio signals from different sources, such as microphones or audio inputs, and analyzes their signal-to-noise ratios (SNR). The system selects or processes only those audio signals where the SNR meets a predefined threshold, ensuring that only high-quality audio signals are used for further processing. This filtering step helps reduce background noise and improve the clarity of the final output. The system may then combine, synchronize, or otherwise process the selected audio signals to produce an enhanced audio output. This approach is particularly useful in environments with varying noise levels, such as conference calls, live broadcasts, or speech recognition applications, where maintaining high audio fidelity is critical. By dynamically selecting audio signals based on SNR, the system ensures that only the most reliable and clear audio inputs are utilized, improving overall audio performance.
11. The media system of claim 1 , wherein a resolution of the first set of audio processing algorithms is lower than a resolution of the second set of audio processing algorithms.
The invention relates to a media system designed to optimize audio processing efficiency by dynamically adjusting the resolution of audio processing algorithms based on system demands. The system includes a processor configured to execute a first set of audio processing algorithms and a second set of audio processing algorithms. The first set operates at a lower resolution compared to the second set, allowing for reduced computational load when high-fidelity processing is not required. The system further includes a memory storing the algorithms and a user interface for selecting audio processing modes. The processor dynamically switches between the first and second sets of algorithms based on factors such as available processing resources, user preferences, or media content characteristics. This adaptive approach ensures efficient resource utilization while maintaining audio quality when needed. The system may also include input and output interfaces for receiving and transmitting audio signals, as well as a power management module to further optimize energy consumption. The invention addresses the challenge of balancing computational efficiency with audio quality in media systems, particularly in devices with limited processing power or battery life.
12. The media system of claim 1 , wherein a signal measure of the combined audio signal is higher than a respective signal measure of any of the set of audio signals or any of the re-processed at least two audio signals.
The media system processes multiple audio signals to enhance audio quality in environments with overlapping or interfering sounds. The system receives a set of audio signals from different sources, such as microphones or audio devices, and applies signal processing techniques to improve clarity and intelligibility. The system combines at least two of these audio signals into a single audio output, where the combined signal exhibits a higher signal measure—such as signal-to-noise ratio, amplitude, or clarity—than any of the individual input signals or any of the re-processed signals. The re-processing may include noise reduction, echo cancellation, or beamforming to isolate desired audio sources. By dynamically adjusting the combination of signals based on their quality metrics, the system ensures that the final output is optimized for listening, even in noisy or complex acoustic environments. This approach is particularly useful in applications like conference calls, live broadcasts, or smart home devices where multiple audio inputs must be merged into a single, high-quality output. The system may also include adaptive filtering and real-time analysis to continuously refine the signal combination for optimal performance.
13. The media system of claim 1 , wherein the signal measure is a signal to noise ratio.
A media system is designed to enhance audio or video signal quality by analyzing and processing input signals. The system includes a signal processing module that evaluates the quality of an incoming media signal, such as audio or video, by measuring a specific signal characteristic. In this particular configuration, the signal measure used is the signal-to-noise ratio (SNR), which quantifies the ratio of the desired signal power to the background noise power. By assessing the SNR, the system can determine the clarity and fidelity of the media signal. The signal processing module may then apply various techniques, such as noise reduction, filtering, or amplification, to improve the signal quality based on the SNR measurement. This ensures that the output media signal is optimized for playback, transmission, or further processing. The system may be integrated into devices like televisions, audio players, or communication systems where signal quality is critical. The use of SNR as the signal measure allows for precise evaluation and enhancement of media signals, addressing issues related to poor audio or video quality due to noise interference.
14. The media system of claim 1 , wherein the network device is one of the first playback device or a server.
A media system is designed to manage and synchronize playback of media content across multiple playback devices in a networked environment. The system addresses challenges in coordinating playback timing, ensuring synchronization, and managing media content distribution among devices. The system includes a network device that controls playback operations, such as starting, pausing, or adjusting playback speed, and ensures that all connected playback devices remain synchronized. The network device may be either one of the playback devices or a dedicated server. When acting as a server, it centrally manages playback commands and synchronization signals, while when integrated into a playback device, it distributes control functions across the network. The system also handles media content storage, streaming, and buffering to minimize latency and ensure smooth playback. The network device may further manage user inputs, such as playback commands or content selection, and relay them to the appropriate devices. This architecture allows for flexible deployment, whether in a peer-to-peer or client-server configuration, while maintaining synchronized playback across all devices. The system is particularly useful in multi-room audio setups, home theater systems, or collaborative media playback environments.
15. A method comprising: receiving, by a first playback device from a second playback device, an indication of an available amount of computational power of the second playback device; and based on the received indication of the available amount of computational power of the second playback device, sending, from the first playback device to the second playback device, a set of audio signals received by one or more microphones of the first playback device wherein the set of audio signals includes at least one voice input; receiving, by the second playback device from the first playback device, the set of audio signals received by the one or more microphones of the first playback device; processing, by the second playback device, the set of audio signals using a first set of audio processing algorithms to determine a set of signal measures corresponding to the set of audio signals; based on the set of signal measures, identifying, by the second playback device from the set of audio signals, at least two audio signals that are to be re-processed using a second set of audio processing algorithms so as to improve the respective signal measures of the at least two audio signals; re-processing, by the second playback device, the at least two audio signals using the second set of audio processing algorithms; combining, by the second playback device, the re-processed at least two audio signals into a combined audio signal; and sending, by the second playback device, the combined audio signal to a network device.
This invention relates to distributed audio processing in a multi-device system, addressing the challenge of efficiently handling voice inputs from microphones in a networked environment. The method involves a first playback device receiving an indication of available computational power from a second playback device. Based on this indication, the first device sends a set of audio signals, including voice inputs captured by its microphones, to the second device. The second device processes these signals using a first set of audio processing algorithms to determine signal measures, such as quality or clarity metrics. It then identifies at least two audio signals that require further processing to improve their signal measures. These signals are re-processed using a second set of algorithms, combined into a single audio signal, and sent to a network device. The approach leverages the computational resources of the second device to enhance audio quality, particularly in scenarios where the first device lacks sufficient processing power. The system dynamically allocates tasks between devices to optimize performance and resource utilization.
16. The method of claim 15 , wherein the network device is one of the first playback device or a server.
A method for managing audio playback in a multi-device network involves synchronizing audio playback across multiple playback devices. The method includes determining a playback position for a first playback device, adjusting the playback position based on a network latency between the first playback device and a second playback device, and transmitting the adjusted playback position to the second playback device. This ensures synchronized playback despite network delays. The method also involves detecting a playback error, such as a buffer underrun, and adjusting the playback position to compensate for the error, maintaining synchronization. Additionally, the method may include dynamically adjusting playback parameters, such as volume or equalization, based on environmental factors or user preferences. The network device responsible for managing this synchronization can be either the first playback device or a central server, depending on the system configuration. This approach improves audio playback consistency in distributed multi-device environments, addressing issues like latency and synchronization errors that degrade user experience.
17. The method of claim 15 , the second playback device further comprising one or more microphones.
A system and method for audio playback synchronization between multiple playback devices, addressing the problem of maintaining consistent audio timing and quality across distributed devices in a networked environment. The system includes a first playback device that generates a synchronization signal and a second playback device that receives and uses this signal to synchronize its audio output with the first device. The synchronization signal is derived from a reference audio signal, ensuring that both devices play audio in unison. The second playback device may also include one or more microphones to capture ambient sound, which can be used for additional synchronization refinement or acoustic feedback analysis. The system dynamically adjusts playback timing based on network latency and device processing delays to minimize audio drift. The method involves transmitting the synchronization signal from the first device to the second, processing the signal at the second device to align playback, and continuously monitoring and adjusting synchronization to maintain alignment. The inclusion of microphones in the second device enables real-time environmental adaptation, improving synchronization accuracy in varying acoustic conditions. This approach ensures seamless, synchronized audio playback across multiple devices, enhancing user experience in multi-device audio systems.
18. The method of claim 15 , further comprising: based on the received indication of the available amount of computational power of the second playback device, determining, by the first playback device, that the available amount of computational power of the second playback device is above a threshold amount of computational power.
This invention relates to distributed audio processing in multi-device playback systems. The problem addressed is efficiently distributing computational tasks between playback devices to optimize performance, particularly when devices have varying computational capabilities. The method involves a first playback device receiving an indication of the available computational power of a second playback device. The first playback device then determines whether the second device's available computational power exceeds a predefined threshold. If the threshold is met, the first device can offload or distribute audio processing tasks to the second device, leveraging its computational resources. This ensures that computationally intensive operations, such as audio decoding, equalization, or synchronization, are handled by the most capable device in the system, improving overall performance and reducing latency. The method may also involve dynamically adjusting task distribution based on real-time computational power assessments, ensuring efficient resource utilization across the networked playback devices. This approach is particularly useful in multi-room audio systems where devices may have different processing capabilities.
19. The method of claim 15 , wherein the at least two audio signals comprise audio signals having signal to noise ratios meeting a threshold amount.
This invention relates to audio signal processing, specifically improving the quality of audio signals by selecting and combining multiple audio signals based on their signal-to-noise ratios (SNR). The problem addressed is the degradation of audio quality in noisy environments, where a single microphone or audio source may capture poor-quality audio due to background noise, interference, or other distortions. The method involves capturing at least two audio signals from different sources, such as multiple microphones or audio inputs. Each audio signal is analyzed to determine its SNR, which measures the ratio of the desired audio signal to unwanted noise. The method then selects audio signals that meet a predefined SNR threshold, ensuring only high-quality signals are used. These selected signals are then combined to produce a final output audio signal with improved clarity and reduced noise. The method may also include additional processing steps, such as filtering, beamforming, or adaptive noise cancellation, to further enhance the combined audio signal. The approach is particularly useful in applications like teleconferencing, speech recognition, and audio recording, where maintaining high audio fidelity in noisy conditions is critical. By dynamically selecting and combining audio signals based on SNR, the method ensures robust and high-quality audio output.
20. The method of claim 15 , wherein the signal measure is a signal to noise ratio.
A system and method for improving signal quality in communication networks addresses the challenge of optimizing signal transmission in environments with interference or noise. The invention involves measuring a signal quality metric to assess the performance of a communication link. Specifically, the signal quality metric is a signal-to-noise ratio (SNR), which quantifies the strength of the desired signal relative to background noise. The system dynamically adjusts transmission parameters, such as power levels or modulation schemes, based on the SNR measurement to enhance signal clarity and reliability. This adaptive approach ensures efficient use of network resources while maintaining high-quality communication. The method may also include comparing the SNR against predefined thresholds to trigger adjustments or alerts, ensuring consistent performance under varying conditions. By focusing on SNR as a key metric, the invention provides a robust solution for optimizing signal integrity in wireless and wired communication systems.
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July 14, 2020
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